similar to: X100P card in Australia

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---------------------------------------------------+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---------------------------------------------------+ Strange things happen under the midnight sun when Men

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages Aug 13 17:23:15 Sheriff kernel: wcfxo: DAA mode is 'FCC' in the file wcfxo.c the following structure is initialised as below which would suggest that FCC is wrong for France or pretty well all of Europe. static struct fxo_mode {

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

I'm new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side. GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying

Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi there, I have a new asterisk working in New Zeland and everything is working well except when an incoming call to the PSTN hangs up, asterisk wont hang up the zap trunk (X100P). I have found this information: http://bugs.digium.com/bug_view_page.php?bug_id=0001474 Which discusses my problem and i have made sure that i have the latest info in the indications.conf as follows: [general]

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

For some thime now I have been trying to get some sense out of Caller ID on PSTN lines in AU and have been getting no where. Now, at last, I seem to be getting something, even though it is an error message (line 5 below). All and any guidance would be welcome. -- Starting simple switch on 'Zap/1-1' Jan 10 09:36:38 NOTICE[18384]: chan_zap.c:5514 ss_thread: Got event 2

Has anyone had problems with Call Waiting signals causing Zap channel or bridging hangups in AU. I was on a call the other day (Zap channel to PSTN) and the call suddenly hung up on my side. I dialled the calling party and got the call again, it seems that the bridge had dropped and that the other party had not lost the connection. As soon as I got the bridging again the other party mentioned